------------------------------
Message: 18
Date: Wed, 15 Mar 2006 11:18:36 +0100
From: Dave Cotton <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Zaptel compile errors on x86_64
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain
On Wed, 2006-03-15 at 17:49 +0800, Walter Klomp wrote:
Hi,
Just downloaded the latest cvs from zaptel on my sparking new Athlon64
Centos4.2 system, but hitting a stumbling block... (sorry for the long
post)
Kernel source installed?
--
Dave Cotton <[EMAIL PROTECTED]>
------------------------------
Message: 19
Date: Wed, 15 Mar 2006 15:19:32 +0500
From: "Mazhar Hussain" <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] There is lacking behind in recorded calls
via sox
To: asterisk-users@lists.digium.com,
[EMAIL PROTECTED]
Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain; charset="iso-8859-1"
Hi ,
I have been using, Monitor (wav, ${CALLFILENAME}) to records calls from
extensions.conf and soxmix software to compiles calls. The problem I am
facing that for long calls more that 2 minutes there is disturbance in
sequence of calls, calls from both ends are not in sequence and there is
always lack behind from one end, I have tried a lot by checking different
version of Sox but still I am facing same issue .Can any one of you will
let me know the reason of this lacking in calls form end after compilation
while calls conversation goes fine in live calls .
Thanks,
Mazhar
Nettechltd.com
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Message: 20
Date: Wed, 15 Mar 2006 15:23:08 +0500
From: "Mazhar Hussain" <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] there is lack behind in recoded calls via
sox
To: [EMAIL PROTECTED],
asterisk-users@lists.digium.com
Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain; charset="iso-8859-1"
Hi ,
I have been using, Monitor (wav, ${CALLFILENAME}) to records calls from
extensions.conf and soxmix software to compiles calls. The problem I am
facing that for long calls more that 2 minutes there is disturbance in
sequence of calls, calls from both ends are not in sequence and there is
always lack behind from one end, I have tried a lot by checking different
version of Sox but still I am facing same issue .Can any one of you will
let me know the reason of this lacking in calls form end after compilation
while calls conversation goes fine in live calls .Also I am using Asterisk
1.2.5 version
Thanks,
Mazhar
Nettechltd.com
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Message: 21
Date: Wed, 15 Mar 2006 11:23:29 +0100
From: "Alejandro Vargas" <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] spa 3000/2100 noise
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1
I've a problem. I've some spa3000 and spa2100. Asterisk 1.2.4.
Prefered codec g711u in both. Calleng from a fxs of spa2100 to the fxo
of spa3000, all works ok. Then I call from a sip phone configured for
using g729, to the fxo of spa3000, it also works ok.
The problem is that after this, when, making again a new call from
spa2100 to spa3000, spa2100 receives only white noise. I suspect a
codec mismatch. The problem disappears by powering off and on the
spa3000.
¿Any ideas on how to check?
--
Alejandro Vargas
------------------------------
Message: 22
Date: Wed, 15 Mar 2006 11:29:37 +0100
From: "Alejandro Vargas" <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Asterisk to receive fax
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1
2006/3/15, Gidean Chan <[EMAIL PROTECTED]>:
Can anyone tell me how to configure my system so that fax can be received
and forward to email account?
You can install iaxfax. It acts as a software modem that connects to
asterisk as a iax phone. It creates a device that can be accesed as a
faxmodem. Then, you can use hylafax that is very powerfull and can be
configured to forward faxes to email, convert it to pdf, etc. etc
(read the documentation).
--
Alejandro Vargas
------------------------------
Message: 23
Date: Wed, 15 Mar 2006 05:34:08 -0500
From: <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Stuck. Extenions.conf? Realtime? MySQL?
Grrrrr!
To: asterisk-users@lists.digium.com
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1
On 3/15/2006, "Douglas Garstang" <[EMAIL PROTECTED]> wrote:
Boy, am I stuck...
[snip]
My brain hurts.
Doug,
Whenever I have gotten to this point in a project, I use two rules for
handling the situation.
Rule 1. Booze
Rule 2. Throw money at it.
Rule 1 makes me feel better.
Rule 2 takes care of the problem but...
If the boss isn't happy - fall back to Rule 1.
The hardest target to hit in the programming shooting gallery is the
moving one. Unless you 'sold' the powers that be that Asterisk is the
answer to all questions... then you made your bed... but as I remember,
I think you got 'stuck' with this one.
You can probably (but I doubt it) buy a system that will do all this
for you. Probably not out of the box tho and probably not without a
large 'programmers' bill to boot. And several third-party packages.
So grab your favorite alcoholic beverage, nail down what they want, and
start solving the problems. Even if it takes a year - it will be better
and cheaper than anything they can purchase.
Brett
------------------------------
Message: 24
Date: Wed, 15 Mar 2006 21:40:41 +1100
From: "James Harper" <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] Re: Cisco phones and Linksys SRW224P
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"
One more thing. Cisco 7905 phone that is working is 74-3092-04 Rev.F0.
Cisco 7905 phone that is not working is 74-3092-08 Rev.A0.
Anybody know about any hardware issue with this revisions?
Nothing for sure, and you may already know this, but some early Cisco
phones only knew how to speak Cisco PoE, not the 802 standard which was
defined a bit later. The Cisco web site should tell you which phone
talks which protocol though.
James
------------------------------
Message: 25
Date: Wed, 15 Mar 2006 11:44:40 +0100
From: Simone Cittadini <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] (unexplicable) peaks of machine load
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-15; format=flowed
I have strange peaks of machine load on my asterisk servers, looking at
top the load is very high even if cpu usage is low and no swap memory is
used.
This happens on all the machines, some of them have asterisk, mysql, agi
and digium cards on them, so I thought I was only asking too much, but
yesterday I noticed the same behaviour on an asterisk machine with only
two digium in it, no other service and a two line extension.
I thought it can be a problem with digium cards but the interrupts
aren't shared, and I have the same problem on a pure-voip server.
Asterisk version varies from 1.2.1 to 1.2.5, the kernels are 2.4 or 2.6
(right ones for the installed cpu, not generic 386)
The only things in common are :
Linux debian, iax channels are used, with jitterbuffer
When this "ghost load" becomes too high (> 3) asterisk starts losing
packets, and the users starts losing patience ...
Anyone experiencing a similar problem ?
------------------------------
Message: 26
Date: Wed, 15 Mar 2006 05:56:20 -0500
From: "Matt Florell" <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] (unexplicable) peaks of machine load
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1
I've noticed this as well from pre 1.0 versions through to 1.2.5
across 12 separate Asterisk servers. The severity seems to be random
mostly. I still haven't figured out what is causing it.
MATT---
On 3/15/06, Simone Cittadini <[EMAIL PROTECTED]> wrote:
I have strange peaks of machine load on my asterisk servers, looking at
top the load is very high even if cpu usage is low and no swap memory is
used.
This happens on all the machines, some of them have asterisk, mysql, agi
and digium cards on them, so I thought I was only asking too much, but
yesterday I noticed the same behaviour on an asterisk machine with only
two digium in it, no other service and a two line extension.
I thought it can be a problem with digium cards but the interrupts
aren't shared, and I have the same problem on a pure-voip server.
Asterisk version varies from 1.2.1 to 1.2.5, the kernels are 2.4 or 2.6
(right ones for the installed cpu, not generic 386)
The only things in common are :
Linux debian, iax channels are used, with jitterbuffer
When this "ghost load" becomes too high (> 3) asterisk starts losing
packets, and the users starts losing patience ...
Anyone experiencing a similar problem ?
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------------------------------
Message: 27
Date: Wed, 15 Mar 2006 22:13:23 +1100
From: James Gardiner <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] CALL FOR COMMENTS - Dialplan
To: asterisk-users@lists.digium.com, asterisk-dev@lists.digium.com
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hello Asterisk community,
I have written a document that covers an Asterisk implementation I am
building.
I want to place it on the lists so USERS can view and make comments on,
the ideas contained within.
I think it is an important issue to develop a standardised Dialplan for
applications, not just for Asterisk, but for pbx systems in general. As
they become cheaper and more common place, each install has its own
ideas of how to implement features.
This makes it very hard for users to move from one system to another.
In any case,
If you have time, please do review the document and make comments to the
list or to me directly.
The document can be found at http://www.crafted.com.au/comments
I cannot post it directly to the list as its TOO BIG.
Thanks,
James
------------------------------
Message: 28
Date: Wed, 15 Mar 2006 22:13:23 +1100
From: James Gardiner <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] [SPAM] [asterisk-dev] CALL FOR COMMENTS -
Dialplan
To: asterisk-users@lists.digium.com, asterisk-dev@lists.digium.com
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hello Asterisk community,
I have written a document that covers an Asterisk implementation I am
building.
I want to place it on the lists so USERS can view and make comments on,
the ideas contained within.
I think it is an important issue to develop a standardised Dialplan for
applications, not just for Asterisk, but for pbx systems in general. As
they become cheaper and more common place, each install has its own
ideas of how to implement features.
This makes it very hard for users to move from one system to another.
In any case,
If you have time, please do review the document and make comments to the
list or to me directly.
The document can be found at http://www.crafted.com.au/comments
I cannot post it directly to the list as its TOO BIG.
Thanks,
James
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------------------------------
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End of Asterisk-Users Digest, Vol 20, Issue 98
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